webrtc/modules/audio_processing/audio_buffer.cc - platform/external/webrtc - Git at Google

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include "webrtc/modules/audio_processing/audio_buffer.h"

 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 namespace webrtc {
 namespace {

 enum {
   kSamplesPer8kHzChannel = 80,
   kSamplesPer16kHzChannel = 160,
   kSamplesPer32kHzChannel = 320
 };

 bool HasKeyboardChannel(AudioProcessing::ChannelLayout layout) {
   switch (layout) {
     case AudioProcessing::kMono:
     case AudioProcessing::kStereo:
       return false;
     case AudioProcessing::kMonoAndKeyboard:
     case AudioProcessing::kStereoAndKeyboard:
       return true;
   }
   assert(false);
   return false;
 }

 int KeyboardChannelIndex(AudioProcessing::ChannelLayout layout) {
   switch (layout) {
     case AudioProcessing::kMono:
     case AudioProcessing::kStereo:
       assert(false);
       return -1;
     case AudioProcessing::kMonoAndKeyboard:
       return 1;
     case AudioProcessing::kStereoAndKeyboard:
       return 2;
   }
   assert(false);
   return -1;
 }

 template <typename T>
 void StereoToMono(const T* left, const T* right, T* out,
                   int samples_per_channel) {
   for (int i = 0; i < samples_per_channel; ++i)
     out[i] = (left[i] + right[i]) / 2;
 }

 }  // namespace

 // One int16_t and one float ChannelBuffer that are kept in sync. The sync is
 // broken when someone requests write access to either ChannelBuffer, and
 // reestablished when someone requests the outdated ChannelBuffer. It is
 // therefore safe to use the return value of ibuf_const() and fbuf_const()
 // until the next call to ibuf() or fbuf(), and the return value of ibuf() and
 // fbuf() until the next call to any of the other functions.
 class IFChannelBuffer {
  public:
   IFChannelBuffer(int samples_per_channel, int num_channels)
       : ivalid_(true),
         ibuf_(samples_per_channel, num_channels),
         fvalid_(true),
         fbuf_(samples_per_channel, num_channels) {}

   ChannelBuffer<int16_t>* ibuf() { return ibuf(false); }
   ChannelBuffer<float>* fbuf() { return fbuf(false); }
   const ChannelBuffer<int16_t>* ibuf_const() { return ibuf(true); }
   const ChannelBuffer<float>* fbuf_const() { return fbuf(true); }

  private:
   ChannelBuffer<int16_t>* ibuf(bool readonly) {
     RefreshI();
     fvalid_ = readonly;
     return &ibuf_;
   }

   ChannelBuffer<float>* fbuf(bool readonly) {
     RefreshF();
     ivalid_ = readonly;
     return &fbuf_;
   }

   void RefreshF() {
     if (!fvalid_) {
       assert(ivalid_);
       const int16_t* const int_data = ibuf_.data();
       float* const float_data = fbuf_.data();
       const int length = fbuf_.length();
       for (int i = 0; i < length; ++i)
         float_data[i] = int_data[i];
       fvalid_ = true;
     }
   }

   void RefreshI() {
     if (!ivalid_) {
       assert(fvalid_);
       FloatS16ToS16(fbuf_.data(), ibuf_.length(), ibuf_.data());
       ivalid_ = true;
     }
   }

   bool ivalid_;
   ChannelBuffer<int16_t> ibuf_;
   bool fvalid_;
   ChannelBuffer<float> fbuf_;
 };

 AudioBuffer::AudioBuffer(int input_samples_per_channel,
                          int num_input_channels,
                          int process_samples_per_channel,
                          int num_process_channels,
                          int output_samples_per_channel)
   : input_samples_per_channel_(input_samples_per_channel),
     num_input_channels_(num_input_channels),
     proc_samples_per_channel_(process_samples_per_channel),
     num_proc_channels_(num_process_channels),
     output_samples_per_channel_(output_samples_per_channel),
     samples_per_split_channel_(proc_samples_per_channel_),
     mixed_low_pass_valid_(false),
     reference_copied_(false),
     activity_(AudioFrame::kVadUnknown),
     keyboard_data_(NULL),
     channels_(new IFChannelBuffer(proc_samples_per_channel_,
                                   num_proc_channels_)) {
   assert(input_samples_per_channel_ > 0);
   assert(proc_samples_per_channel_ > 0);
   assert(output_samples_per_channel_ > 0);
   assert(num_input_channels_ > 0 && num_input_channels_ <= 2);
   assert(num_proc_channels_ <= num_input_channels);

   if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
     input_buffer_.reset(new ChannelBuffer<float>(input_samples_per_channel_,
                                                  num_proc_channels_));
   }

   if (input_samples_per_channel_ != proc_samples_per_channel_ ||
       output_samples_per_channel_ != proc_samples_per_channel_) {
     // Create an intermediate buffer for resampling.
     process_buffer_.reset(new ChannelBuffer<float>(proc_samples_per_channel_,
                                                    num_proc_channels_));
   }

   if (input_samples_per_channel_ != proc_samples_per_channel_) {
     input_resamplers_.reserve(num_proc_channels_);
     for (int i = 0; i < num_proc_channels_; ++i) {
       input_resamplers_.push_back(
           new PushSincResampler(input_samples_per_channel_,
                                 proc_samples_per_channel_));
     }
   }

   if (output_samples_per_channel_ != proc_samples_per_channel_) {
     output_resamplers_.reserve(num_proc_channels_);
     for (int i = 0; i < num_proc_channels_; ++i) {
       output_resamplers_.push_back(
           new PushSincResampler(proc_samples_per_channel_,
                                 output_samples_per_channel_));
     }
   }

   if (proc_samples_per_channel_ == kSamplesPer32kHzChannel) {
     samples_per_split_channel_ = kSamplesPer16kHzChannel;
     split_channels_low_.reset(new IFChannelBuffer(samples_per_split_channel_,
                                                   num_proc_channels_));
     split_channels_high_.reset(new IFChannelBuffer(samples_per_split_channel_,
                                                    num_proc_channels_));
     filter_states_.reset(new SplitFilterStates[num_proc_channels_]);
   }
 }

 AudioBuffer::~AudioBuffer() {}

 void AudioBuffer::CopyFrom(const float* const* data,
                            int samples_per_channel,
                            AudioProcessing::ChannelLayout layout) {
   assert(samples_per_channel == input_samples_per_channel_);
   assert(ChannelsFromLayout(layout) == num_input_channels_);
   InitForNewData();

   if (HasKeyboardChannel(layout)) {
     keyboard_data_ = data[KeyboardChannelIndex(layout)];
   }

   // Downmix.
   const float* const* data_ptr = data;
   if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
     StereoToMono(data[0],
                  data[1],
                  input_buffer_->channel(0),
                  input_samples_per_channel_);
     data_ptr = input_buffer_->channels();
   }

   // Resample.
   if (input_samples_per_channel_ != proc_samples_per_channel_) {
     for (int i = 0; i < num_proc_channels_; ++i) {
       input_resamplers_[i]->Resample(data_ptr[i],
                                      input_samples_per_channel_,
                                      process_buffer_->channel(i),
                                      proc_samples_per_channel_);
     }
     data_ptr = process_buffer_->channels();
   }

   // Convert to int16.
   for (int i = 0; i < num_proc_channels_; ++i) {
     FloatToFloatS16(data_ptr[i], proc_samples_per_channel_,
                 channels_->fbuf()->channel(i));
   }
 }

 void AudioBuffer::CopyTo(int samples_per_channel,
                          AudioProcessing::ChannelLayout layout,
                          float* const* data) {
   assert(samples_per_channel == output_samples_per_channel_);
   assert(ChannelsFromLayout(layout) == num_proc_channels_);

   // Convert to float.
   float* const* data_ptr = data;
   if (output_samples_per_channel_ != proc_samples_per_channel_) {
     // Convert to an intermediate buffer for subsequent resampling.
     data_ptr = process_buffer_->channels();
   }
   for (int i = 0; i < num_proc_channels_; ++i) {
     FloatS16ToFloat(channels_->fbuf()->channel(i),
                 proc_samples_per_channel_,
                 data_ptr[i]);
   }

   // Resample.
   if (output_samples_per_channel_ != proc_samples_per_channel_) {
     for (int i = 0; i < num_proc_channels_; ++i) {
       output_resamplers_[i]->Resample(data_ptr[i],
                                       proc_samples_per_channel_,
                                       data[i],
                                       output_samples_per_channel_);
     }
   }
 }

 void AudioBuffer::InitForNewData() {
   keyboard_data_ = NULL;
   mixed_low_pass_valid_ = false;
   reference_copied_ = false;
   activity_ = AudioFrame::kVadUnknown;
 }

 const int16_t* AudioBuffer::data(int channel) const {
   return channels_->ibuf_const()->channel(channel);
 }

 int16_t* AudioBuffer::data(int channel) {
   mixed_low_pass_valid_ = false;
   return channels_->ibuf()->channel(channel);
 }

 const float* AudioBuffer::data_f(int channel) const {
   return channels_->fbuf_const()->channel(channel);
 }

 float* AudioBuffer::data_f(int channel) {
   mixed_low_pass_valid_ = false;
   return channels_->fbuf()->channel(channel);
 }

 const float* const* AudioBuffer::channels_f() const {
   return channels_->fbuf_const()->channels();
 }

 float* const* AudioBuffer::channels_f() {
   mixed_low_pass_valid_ = false;
   return channels_->fbuf()->channels();
 }

 const int16_t* AudioBuffer::low_pass_split_data(int channel) const {
   return split_channels_low_.get()
       ? split_channels_low_->ibuf_const()->channel(channel)
       : data(channel);
 }

 int16_t* AudioBuffer::low_pass_split_data(int channel) {
   mixed_low_pass_valid_ = false;
   return split_channels_low_.get()
       ? split_channels_low_->ibuf()->channel(channel)
       : data(channel);
 }

 const float* AudioBuffer::low_pass_split_data_f(int channel) const {
   return split_channels_low_.get()
       ? split_channels_low_->fbuf_const()->channel(channel)
       : data_f(channel);
 }

 float* AudioBuffer::low_pass_split_data_f(int channel) {
   mixed_low_pass_valid_ = false;
   return split_channels_low_.get()
       ? split_channels_low_->fbuf()->channel(channel)
       : data_f(channel);
 }

 const float* const* AudioBuffer::low_pass_split_channels_f() const {
   return split_channels_low_.get()
       ? split_channels_low_->fbuf_const()->channels()
       : channels_f();
 }

 float* const* AudioBuffer::low_pass_split_channels_f() {
   mixed_low_pass_valid_ = false;
   return split_channels_low_.get()
       ? split_channels_low_->fbuf()->channels()
       : channels_f();
 }

 const int16_t* AudioBuffer::high_pass_split_data(int channel) const {
   return split_channels_high_.get()
       ? split_channels_high_->ibuf_const()->channel(channel)
       : NULL;
 }

 int16_t* AudioBuffer::high_pass_split_data(int channel) {
   return split_channels_high_.get()
       ? split_channels_high_->ibuf()->channel(channel)
       : NULL;
 }

 const float* AudioBuffer::high_pass_split_data_f(int channel) const {
   return split_channels_high_.get()
       ? split_channels_high_->fbuf_const()->channel(channel)
       : NULL;
 }

 float* AudioBuffer::high_pass_split_data_f(int channel) {
   return split_channels_high_.get()
       ? split_channels_high_->fbuf()->channel(channel)
       : NULL;
 }

 const float* const* AudioBuffer::high_pass_split_channels_f() const {
   return split_channels_high_.get()
       ? split_channels_high_->fbuf_const()->channels()
       : NULL;
 }

 float* const* AudioBuffer::high_pass_split_channels_f() {
   return split_channels_high_.get()
       ? split_channels_high_->fbuf()->channels()
       : NULL;
 }

 const int16_t* AudioBuffer::mixed_low_pass_data() {
   // Currently only mixing stereo to mono is supported.
   assert(num_proc_channels_ == 1 || num_proc_channels_ == 2);

   if (num_proc_channels_ == 1) {
     return low_pass_split_data(0);
   }

   if (!mixed_low_pass_valid_) {
     if (!mixed_low_pass_channels_.get()) {
       mixed_low_pass_channels_.reset(
           new ChannelBuffer<int16_t>(samples_per_split_channel_, 1));
     }
     StereoToMono(low_pass_split_data(0),
                  low_pass_split_data(1),
                  mixed_low_pass_channels_->data(),
                  samples_per_split_channel_);
     mixed_low_pass_valid_ = true;
   }
   return mixed_low_pass_channels_->data();
 }

 const int16_t* AudioBuffer::low_pass_reference(int channel) const {
   if (!reference_copied_) {
     return NULL;
   }

   return low_pass_reference_channels_->channel(channel);
 }

 const float* AudioBuffer::keyboard_data() const {
   return keyboard_data_;
 }

 SplitFilterStates* AudioBuffer::filter_states(int channel) {
   assert(channel >= 0 && channel < num_proc_channels_);
   return &filter_states_[channel];
 }

 void AudioBuffer::set_activity(AudioFrame::VADActivity activity) {
   activity_ = activity;
 }

 AudioFrame::VADActivity AudioBuffer::activity() const {
   return activity_;
 }

 int AudioBuffer::num_channels() const {
   return num_proc_channels_;
 }

 int AudioBuffer::samples_per_channel() const {
   return proc_samples_per_channel_;
 }

 int AudioBuffer::samples_per_split_channel() const {
   return samples_per_split_channel_;
 }

 int AudioBuffer::samples_per_keyboard_channel() const {
   // We don't resample the keyboard channel.
   return input_samples_per_channel_;
 }

 // TODO(andrew): Do deinterleaving and mixing in one step?
 void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
   assert(proc_samples_per_channel_ == input_samples_per_channel_);
   assert(frame->num_channels_ == num_input_channels_);
   assert(frame->samples_per_channel_ ==  proc_samples_per_channel_);
   InitForNewData();
   activity_ = frame->vad_activity_;

   if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
     // Downmix directly; no explicit deinterleaving needed.
     int16_t* downmixed = channels_->ibuf()->channel(0);
     for (int i = 0; i < input_samples_per_channel_; ++i) {
       downmixed[i] = (frame->data_[i * 2] + frame->data_[i * 2 + 1]) / 2;
     }
   } else {
     assert(num_proc_channels_ == num_input_channels_);
     int16_t* interleaved = frame->data_;
     for (int i = 0; i < num_proc_channels_; ++i) {
       int16_t* deinterleaved = channels_->ibuf()->channel(i);
       int interleaved_idx = i;
       for (int j = 0; j < proc_samples_per_channel_; ++j) {
         deinterleaved[j] = interleaved[interleaved_idx];
         interleaved_idx += num_proc_channels_;
       }
     }
   }
 }

 void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const {
   assert(proc_samples_per_channel_ == output_samples_per_channel_);
   assert(num_proc_channels_ == num_input_channels_);
   assert(frame->num_channels_ == num_proc_channels_);
   assert(frame->samples_per_channel_ == proc_samples_per_channel_);
   frame->vad_activity_ = activity_;

   if (!data_changed) {
     return;
   }

   int16_t* interleaved = frame->data_;
   for (int i = 0; i < num_proc_channels_; i++) {
     int16_t* deinterleaved = channels_->ibuf()->channel(i);
     int interleaved_idx = i;
     for (int j = 0; j < proc_samples_per_channel_; j++) {
       interleaved[interleaved_idx] = deinterleaved[j];
       interleaved_idx += num_proc_channels_;
     }
   }
 }

 void AudioBuffer::CopyLowPassToReference() {
   reference_copied_ = true;
   if (!low_pass_reference_channels_.get()) {
     low_pass_reference_channels_.reset(
         new ChannelBuffer<int16_t>(samples_per_split_channel_,
                                    num_proc_channels_));
   }
   for (int i = 0; i < num_proc_channels_; i++) {
     low_pass_reference_channels_->CopyFrom(low_pass_split_data(i), i);
   }
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include "webrtc/modules/audio_processing/audio_buffer.h"

	#include "webrtc/common_audio/include/audio_util.h"
	#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
	#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

	namespace webrtc {
	namespace {

	enum {
	kSamplesPer8kHzChannel = 80,
	kSamplesPer16kHzChannel = 160,
	kSamplesPer32kHzChannel = 320
	};

	bool HasKeyboardChannel(AudioProcessing::ChannelLayout layout) {
	switch (layout) {
	case AudioProcessing::kMono:
	case AudioProcessing::kStereo:
	return false;
	case AudioProcessing::kMonoAndKeyboard:
	case AudioProcessing::kStereoAndKeyboard:
	return true;
	}
	assert(false);
	return false;
	}

	int KeyboardChannelIndex(AudioProcessing::ChannelLayout layout) {
	switch (layout) {
	case AudioProcessing::kMono:
	case AudioProcessing::kStereo:
	assert(false);
	return -1;
	case AudioProcessing::kMonoAndKeyboard:
	return 1;
	case AudioProcessing::kStereoAndKeyboard:
	return 2;
	}
	assert(false);
	return -1;
	}

	template <typename T>
	void StereoToMono(const T* left, const T* right, T* out,
	int samples_per_channel) {
	for (int i = 0; i < samples_per_channel; ++i)
	out[i] = (left[i] + right[i]) / 2;
	}

	} // namespace

	// One int16_t and one float ChannelBuffer that are kept in sync. The sync is
	// broken when someone requests write access to either ChannelBuffer, and
	// reestablished when someone requests the outdated ChannelBuffer. It is
	// therefore safe to use the return value of ibuf_const() and fbuf_const()
	// until the next call to ibuf() or fbuf(), and the return value of ibuf() and
	// fbuf() until the next call to any of the other functions.
	class IFChannelBuffer {
	public:
	IFChannelBuffer(int samples_per_channel, int num_channels)
	: ivalid_(true),
	ibuf_(samples_per_channel, num_channels),
	fvalid_(true),
	fbuf_(samples_per_channel, num_channels) {}

	ChannelBuffer<int16_t>* ibuf() { return ibuf(false); }
	ChannelBuffer<float>* fbuf() { return fbuf(false); }
	const ChannelBuffer<int16_t>* ibuf_const() { return ibuf(true); }
	const ChannelBuffer<float>* fbuf_const() { return fbuf(true); }

	private:
	ChannelBuffer<int16_t>* ibuf(bool readonly) {
	RefreshI();
	fvalid_ = readonly;
	return &ibuf_;
	}

	ChannelBuffer<float>* fbuf(bool readonly) {
	RefreshF();
	ivalid_ = readonly;
	return &fbuf_;
	}

	void RefreshF() {
	if (!fvalid_) {
	assert(ivalid_);
	const int16_t* const int_data = ibuf_.data();
	float* const float_data = fbuf_.data();
	const int length = fbuf_.length();
	for (int i = 0; i < length; ++i)
	float_data[i] = int_data[i];
	fvalid_ = true;
	}
	}

	void RefreshI() {
	if (!ivalid_) {
	assert(fvalid_);
	FloatS16ToS16(fbuf_.data(), ibuf_.length(), ibuf_.data());
	ivalid_ = true;
	}
	}

	bool ivalid_;
	ChannelBuffer<int16_t> ibuf_;
	bool fvalid_;
	ChannelBuffer<float> fbuf_;
	};

	AudioBuffer::AudioBuffer(int input_samples_per_channel,
	int num_input_channels,
	int process_samples_per_channel,
	int num_process_channels,
	int output_samples_per_channel)
	: input_samples_per_channel_(input_samples_per_channel),
	num_input_channels_(num_input_channels),
	proc_samples_per_channel_(process_samples_per_channel),
	num_proc_channels_(num_process_channels),
	output_samples_per_channel_(output_samples_per_channel),
	samples_per_split_channel_(proc_samples_per_channel_),
	mixed_low_pass_valid_(false),
	reference_copied_(false),
	activity_(AudioFrame::kVadUnknown),
	keyboard_data_(NULL),
	channels_(new IFChannelBuffer(proc_samples_per_channel_,
	num_proc_channels_)) {
	assert(input_samples_per_channel_ > 0);
	assert(proc_samples_per_channel_ > 0);
	assert(output_samples_per_channel_ > 0);
	assert(num_input_channels_ > 0 && num_input_channels_ <= 2);
	assert(num_proc_channels_ <= num_input_channels);

	if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
	input_buffer_.reset(new ChannelBuffer<float>(input_samples_per_channel_,
	num_proc_channels_));
	}

	if (input_samples_per_channel_ != proc_samples_per_channel_ \|\|
	output_samples_per_channel_ != proc_samples_per_channel_) {
	// Create an intermediate buffer for resampling.
	process_buffer_.reset(new ChannelBuffer<float>(proc_samples_per_channel_,
	num_proc_channels_));
	}

	if (input_samples_per_channel_ != proc_samples_per_channel_) {
	input_resamplers_.reserve(num_proc_channels_);
	for (int i = 0; i < num_proc_channels_; ++i) {
	input_resamplers_.push_back(
	new PushSincResampler(input_samples_per_channel_,
	proc_samples_per_channel_));
	}
	}

	if (output_samples_per_channel_ != proc_samples_per_channel_) {
	output_resamplers_.reserve(num_proc_channels_);
	for (int i = 0; i < num_proc_channels_; ++i) {
	output_resamplers_.push_back(
	new PushSincResampler(proc_samples_per_channel_,
	output_samples_per_channel_));
	}
	}

	if (proc_samples_per_channel_ == kSamplesPer32kHzChannel) {
	samples_per_split_channel_ = kSamplesPer16kHzChannel;
	split_channels_low_.reset(new IFChannelBuffer(samples_per_split_channel_,
	num_proc_channels_));
	split_channels_high_.reset(new IFChannelBuffer(samples_per_split_channel_,
	num_proc_channels_));
	filter_states_.reset(new SplitFilterStates[num_proc_channels_]);
	}
	}

	AudioBuffer::~AudioBuffer() {}

	void AudioBuffer::CopyFrom(const float* const* data,
	int samples_per_channel,
	AudioProcessing::ChannelLayout layout) {
	assert(samples_per_channel == input_samples_per_channel_);
	assert(ChannelsFromLayout(layout) == num_input_channels_);
	InitForNewData();

	if (HasKeyboardChannel(layout)) {
	keyboard_data_ = data[KeyboardChannelIndex(layout)];
	}

	// Downmix.
	const float* const* data_ptr = data;
	if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
	StereoToMono(data[0],
	data[1],
	input_buffer_->channel(0),
	input_samples_per_channel_);
	data_ptr = input_buffer_->channels();
	}

	// Resample.
	if (input_samples_per_channel_ != proc_samples_per_channel_) {
	for (int i = 0; i < num_proc_channels_; ++i) {
	input_resamplers_[i]->Resample(data_ptr[i],
	input_samples_per_channel_,
	process_buffer_->channel(i),
	proc_samples_per_channel_);
	}
	data_ptr = process_buffer_->channels();
	}

	// Convert to int16.
	for (int i = 0; i < num_proc_channels_; ++i) {
	FloatToFloatS16(data_ptr[i], proc_samples_per_channel_,
	channels_->fbuf()->channel(i));
	}
	}

	void AudioBuffer::CopyTo(int samples_per_channel,
	AudioProcessing::ChannelLayout layout,
	float* const* data) {
	assert(samples_per_channel == output_samples_per_channel_);
	assert(ChannelsFromLayout(layout) == num_proc_channels_);

	// Convert to float.
	float* const* data_ptr = data;
	if (output_samples_per_channel_ != proc_samples_per_channel_) {
	// Convert to an intermediate buffer for subsequent resampling.
	data_ptr = process_buffer_->channels();
	}
	for (int i = 0; i < num_proc_channels_; ++i) {
	FloatS16ToFloat(channels_->fbuf()->channel(i),
	proc_samples_per_channel_,
	data_ptr[i]);
	}

	// Resample.
	if (output_samples_per_channel_ != proc_samples_per_channel_) {
	for (int i = 0; i < num_proc_channels_; ++i) {
	output_resamplers_[i]->Resample(data_ptr[i],
	proc_samples_per_channel_,
	data[i],
	output_samples_per_channel_);
	}
	}
	}

	void AudioBuffer::InitForNewData() {
	keyboard_data_ = NULL;
	mixed_low_pass_valid_ = false;
	reference_copied_ = false;
	activity_ = AudioFrame::kVadUnknown;
	}

	const int16_t* AudioBuffer::data(int channel) const {
	return channels_->ibuf_const()->channel(channel);
	}

	int16_t* AudioBuffer::data(int channel) {
	mixed_low_pass_valid_ = false;
	return channels_->ibuf()->channel(channel);
	}

	const float* AudioBuffer::data_f(int channel) const {
	return channels_->fbuf_const()->channel(channel);
	}

	float* AudioBuffer::data_f(int channel) {
	mixed_low_pass_valid_ = false;
	return channels_->fbuf()->channel(channel);
	}

	const float* const* AudioBuffer::channels_f() const {
	return channels_->fbuf_const()->channels();
	}

	float* const* AudioBuffer::channels_f() {
	mixed_low_pass_valid_ = false;
	return channels_->fbuf()->channels();
	}

	const int16_t* AudioBuffer::low_pass_split_data(int channel) const {
	return split_channels_low_.get()
	? split_channels_low_->ibuf_const()->channel(channel)
	: data(channel);
	}

	int16_t* AudioBuffer::low_pass_split_data(int channel) {
	mixed_low_pass_valid_ = false;
	return split_channels_low_.get()
	? split_channels_low_->ibuf()->channel(channel)
	: data(channel);
	}

	const float* AudioBuffer::low_pass_split_data_f(int channel) const {
	return split_channels_low_.get()
	? split_channels_low_->fbuf_const()->channel(channel)
	: data_f(channel);
	}

	float* AudioBuffer::low_pass_split_data_f(int channel) {
	mixed_low_pass_valid_ = false;
	return split_channels_low_.get()
	? split_channels_low_->fbuf()->channel(channel)
	: data_f(channel);
	}

	const float* const* AudioBuffer::low_pass_split_channels_f() const {
	return split_channels_low_.get()
	? split_channels_low_->fbuf_const()->channels()
	: channels_f();
	}

	float* const* AudioBuffer::low_pass_split_channels_f() {
	mixed_low_pass_valid_ = false;
	return split_channels_low_.get()
	? split_channels_low_->fbuf()->channels()
	: channels_f();
	}

	const int16_t* AudioBuffer::high_pass_split_data(int channel) const {
	return split_channels_high_.get()
	? split_channels_high_->ibuf_const()->channel(channel)
	: NULL;
	}

	int16_t* AudioBuffer::high_pass_split_data(int channel) {
	return split_channels_high_.get()
	? split_channels_high_->ibuf()->channel(channel)
	: NULL;
	}

	const float* AudioBuffer::high_pass_split_data_f(int channel) const {
	return split_channels_high_.get()
	? split_channels_high_->fbuf_const()->channel(channel)
	: NULL;
	}

	float* AudioBuffer::high_pass_split_data_f(int channel) {
	return split_channels_high_.get()
	? split_channels_high_->fbuf()->channel(channel)
	: NULL;
	}

	const float* const* AudioBuffer::high_pass_split_channels_f() const {
	return split_channels_high_.get()
	? split_channels_high_->fbuf_const()->channels()
	: NULL;
	}

	float* const* AudioBuffer::high_pass_split_channels_f() {
	return split_channels_high_.get()
	? split_channels_high_->fbuf()->channels()
	: NULL;
	}

	const int16_t* AudioBuffer::mixed_low_pass_data() {
	// Currently only mixing stereo to mono is supported.
	assert(num_proc_channels_ == 1 \|\| num_proc_channels_ == 2);

	if (num_proc_channels_ == 1) {
	return low_pass_split_data(0);
	}

	if (!mixed_low_pass_valid_) {
	if (!mixed_low_pass_channels_.get()) {
	mixed_low_pass_channels_.reset(
	new ChannelBuffer<int16_t>(samples_per_split_channel_, 1));
	}
	StereoToMono(low_pass_split_data(0),
	low_pass_split_data(1),
	mixed_low_pass_channels_->data(),
	samples_per_split_channel_);
	mixed_low_pass_valid_ = true;
	}
	return mixed_low_pass_channels_->data();
	}

	const int16_t* AudioBuffer::low_pass_reference(int channel) const {
	if (!reference_copied_) {
	return NULL;
	}

	return low_pass_reference_channels_->channel(channel);
	}

	const float* AudioBuffer::keyboard_data() const {
	return keyboard_data_;
	}

	SplitFilterStates* AudioBuffer::filter_states(int channel) {
	assert(channel >= 0 && channel < num_proc_channels_);
	return &filter_states_[channel];
	}

	void AudioBuffer::set_activity(AudioFrame::VADActivity activity) {
	activity_ = activity;
	}

	AudioFrame::VADActivity AudioBuffer::activity() const {
	return activity_;
	}

	int AudioBuffer::num_channels() const {
	return num_proc_channels_;
	}

	int AudioBuffer::samples_per_channel() const {
	return proc_samples_per_channel_;
	}

	int AudioBuffer::samples_per_split_channel() const {
	return samples_per_split_channel_;
	}

	int AudioBuffer::samples_per_keyboard_channel() const {
	// We don't resample the keyboard channel.
	return input_samples_per_channel_;
	}

	// TODO(andrew): Do deinterleaving and mixing in one step?
	void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
	assert(proc_samples_per_channel_ == input_samples_per_channel_);
	assert(frame->num_channels_ == num_input_channels_);
	assert(frame->samples_per_channel_ == proc_samples_per_channel_);
	InitForNewData();
	activity_ = frame->vad_activity_;

	if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
	// Downmix directly; no explicit deinterleaving needed.
	int16_t* downmixed = channels_->ibuf()->channel(0);
	for (int i = 0; i < input_samples_per_channel_; ++i) {
	downmixed[i] = (frame->data_[i * 2] + frame->data_[i * 2 + 1]) / 2;
	}
	} else {
	assert(num_proc_channels_ == num_input_channels_);
	int16_t* interleaved = frame->data_;
	for (int i = 0; i < num_proc_channels_; ++i) {
	int16_t* deinterleaved = channels_->ibuf()->channel(i);
	int interleaved_idx = i;
	for (int j = 0; j < proc_samples_per_channel_; ++j) {
	deinterleaved[j] = interleaved[interleaved_idx];
	interleaved_idx += num_proc_channels_;
	}
	}
	}
	}

	void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const {
	assert(proc_samples_per_channel_ == output_samples_per_channel_);
	assert(num_proc_channels_ == num_input_channels_);
	assert(frame->num_channels_ == num_proc_channels_);
	assert(frame->samples_per_channel_ == proc_samples_per_channel_);
	frame->vad_activity_ = activity_;

	if (!data_changed) {
	return;
	}

	int16_t* interleaved = frame->data_;
	for (int i = 0; i < num_proc_channels_; i++) {
	int16_t* deinterleaved = channels_->ibuf()->channel(i);
	int interleaved_idx = i;
	for (int j = 0; j < proc_samples_per_channel_; j++) {
	interleaved[interleaved_idx] = deinterleaved[j];
	interleaved_idx += num_proc_channels_;
	}
	}
	}

	void AudioBuffer::CopyLowPassToReference() {
	reference_copied_ = true;
	if (!low_pass_reference_channels_.get()) {
	low_pass_reference_channels_.reset(
	new ChannelBuffer<int16_t>(samples_per_split_channel_,
	num_proc_channels_));
	}
	for (int i = 0; i < num_proc_channels_; i++) {
	low_pass_reference_channels_->CopyFrom(low_pass_split_data(i), i);
	}
	}

	} // namespace webrtc